In article writes:

What's funny about Nashville Skyline is that Bob Johnston, the album's
producer, used to tell singers to mispronounce the lyrics. He apparantly
took the total opposide approach with Bob Dylan!

Nothing like some good old mounting music. g


--
I'm really Mike Rivers )
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me he double-m-eleven-double-zero at yahoo

In article writes:

The other nice feature with quality gear is that when you go to sell
it, it's easy to sell, you will have lots of people interested in it.

Not where I live, in a minor city at the southern tip of Africa. It's
easy to peddle a used Behringer compressor or some such thing, because
it's cheap, but try and sell anything for more than the equivalent of
about $300 and your potential market shrinks alarmingly.

We have a global economy now. For some people, it may not be a
hardship to pay for shipping from Africa for the the right piece of
gear. eBay is your friend if you're a seller, and occasionally if
you're a buyer.


--
I'm really Mike Rivers )
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me he double-m-eleven-double-zero at yahoo

In article writes:

The other nice feature with quality gear is that when you go to sell
it, it's easy to sell, you will have lots of people interested in it.

Not where I live, in a minor city at the southern tip of Africa. It's
easy to peddle a used Behringer compressor or some such thing, because
it's cheap, but try and sell anything for more than the equivalent of
about $300 and your potential market shrinks alarmingly.

We have a global economy now. For some people, it may not be a
hardship to pay for shipping from Africa for the the right piece of
gear. eBay is your friend if you're a seller, and occasionally if
you're a buyer.


--
I'm really Mike Rivers )
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me he double-m-eleven-double-zero at yahoo

"Buster Mudd" wrote in message

Kalman Rubinson wrote in message
. ..
In article
writes:

Before i forget it, does anyone have thoughts about doing vocals
while monitoring with headphones?

IME, the critical ingredient is the monitoring mix - what the headphones
tell you while you are performing.

I'm chasing down some anecdotal
information that suggests that monitoring with only the right
earcup (which ends up on the left side of the brain) results in
singers being more on pitch.

Before this thread goes any further, let me point out that the basic
premise, that the sound from each ear ends up in the other side of
the brain, is untrue. The input from each ear is relayed to a
number ob sites in the brainstem and, after the first relay, is
represented on BOTH sides of the brain. That certainly holds true
for the cerebral hemispheres; they are binaural.

Excellent point.

Maybe that explains why I hate tracking or overdubbing with one
headphone earpiece on & the other off; I'm much more comfortable with
both earpieces halfway on (raised up 1" or so) if I need to hear
acoustic room sound.

I think that could be true, but it's something different - basically Buster
you've criticized and indicted your monitoring mix. There's something in the
room you need that it does not include - perhaps the sound of your voice as
*processed* by the room and your HRTF.

"Buster Mudd" wrote in message

Kalman Rubinson wrote in message
. ..
In article
writes:

Before i forget it, does anyone have thoughts about doing vocals
while monitoring with headphones?

IME, the critical ingredient is the monitoring mix - what the headphones
tell you while you are performing.

I'm chasing down some anecdotal
information that suggests that monitoring with only the right
earcup (which ends up on the left side of the brain) results in
singers being more on pitch.

Before this thread goes any further, let me point out that the basic
premise, that the sound from each ear ends up in the other side of
the brain, is untrue. The input from each ear is relayed to a
number ob sites in the brainstem and, after the first relay, is
represented on BOTH sides of the brain. That certainly holds true
for the cerebral hemispheres; they are binaural.

Excellent point.

Maybe that explains why I hate tracking or overdubbing with one
headphone earpiece on & the other off; I'm much more comfortable with
both earpieces halfway on (raised up 1" or so) if I need to hear
acoustic room sound.

I think that could be true, but it's something different - basically Buster
you've criticized and indicted your monitoring mix. There's something in the
room you need that it does not include - perhaps the sound of your voice as
*processed* by the room and your HRTF.

On Tue, 07 Sep 2004 17:33:00 -0700, Bob Cain wrote:



philicorda wrote:


I think they took impulse responses of the mics in an anechoic chamber.

That's what they say bu I have my doubts. It's easy to
extract the IR's they use and I've done that for all their
mics and the inverses. When I've compared the magnitude
response of these to published curves for various mics and
they look way more alike than the mics themselves usually do.

I remember reading somewhere that they also got impulses to the
capsules of the condenser mics by connecting the signal directly to the
mic diaphram. I can't see how that would work, other than to get the eq of
the mic electronics. Or perhaps they ran the capsule 'backwards' as an
electrostatic speaker while recording it's output at the same time?

Can't say either why it would have any relevance.


I'm pretty sure it's a convolver with mic impulse responses.. any other
way would be a very long way around.

That's it. I don't think they use FIR's though. I have
reason to think they are mapping the response curves,
however they obtain them, to minimum phase IIR's.

Do you think they use IIRs to keep latency down? My understanding of such
things is a little vague.



I can't say I've ever seen a graph that came with a mic that contains any
detail whatsoever. They tend to have about a +-10db range and a thick
black line.

And you'll see pretty much the same curve if you extract one
of the Antares impulse responses and transform it. Same
smoothness, same shape as published but no thick black line.
:-)

Having a look at a .amm file in a hex editor, the part that
deals with one pattern of a mic, seems to be about 200-350 samples long
(If they are IRs with 16bit samples, they don't look like it, but the
organisation of the file is quite clear). Is that long enough for an mic
impulse response?

I wonder if the IRs are rather generated 'on the fly' from a splined set
of co-ordinates in that file... That would explain the smoothness.




Based on actual measurement I can make my Sony ECM MS-907 mic sound so
much like a pair of Sound Room MC012 cards in XY that I'd almost be
willing to put them up in ABX. Not quite that confident, though, and
it is well nigh impossible to set up an ABX for microphones. That's
part of what makes it so easy for some to say just about anything they
like about mics and classify their quality by how much they cost.


Move off axis, or work in anything but an anechoic chamber, and all
bets are off.... The patterns won't be the same. Also, put them over a
kit as overheads and I bet they sound totally different, if only
because they will crap out in different ways. (The pads on the
octavas are not very nice). Then try a pair of KM84s instead of the
octavas...

Since real life recording situation variations are at least as great and
usually far greater than the differences in patterns between the
microphones, what I am saying is that what I get out of a transformed
ECM-MS907 is a _very_ plausible MC012 XY.

If you want to loan me a pair of KM84's and a Sony ECM-MS907 for about a
week, I'll send them back to you with IR's that will transform MS907
recordings to a plausible facsimile of KM84 recordings that you can
judge for yourself.

If you decide to take me up on that, I will need a bit of setup time
first to get my new hemi-anechoic measurement rig rung out. Pics he

http://www.arcanemethods.com/Hemi_Anechoic/

The thing will be set up out of doors far from reflections.

Cool. I'd guess that knowing the response of an exact set of
source and modelled mics would work rather better than generic
examples.

Any particular reason for using a ECM-MS907?
Also, what's the driver at the bottom of that strange device?
And... What convolver(s) do you use afterwards? I have Acoustic Mirror and
SIR.

Sadly, I can't loan you a pair of KM84s, as I don't own any.
I tend to work on a beg and borrow basis to supplement my meagre
collection of mics. They do sound nicer than my octavas when I've
got them.

It's an interesting offer though. While I 'know' it should not be possible
to turn an ECM-MS907 into a passable pair of KM84s, I also 'knew' that no
softsynth could ever sound fat like a moog, and that all digital eqs sound
nasty and harsh.


Bob

On Tue, 07 Sep 2004 17:33:00 -0700, Bob Cain wrote:



philicorda wrote:


I think they took impulse responses of the mics in an anechoic chamber.

That's what they say bu I have my doubts. It's easy to
extract the IR's they use and I've done that for all their
mics and the inverses. When I've compared the magnitude
response of these to published curves for various mics and
they look way more alike than the mics themselves usually do.

I remember reading somewhere that they also got impulses to the
capsules of the condenser mics by connecting the signal directly to the
mic diaphram. I can't see how that would work, other than to get the eq of
the mic electronics. Or perhaps they ran the capsule 'backwards' as an
electrostatic speaker while recording it's output at the same time?

Can't say either why it would have any relevance.


I'm pretty sure it's a convolver with mic impulse responses.. any other
way would be a very long way around.

That's it. I don't think they use FIR's though. I have
reason to think they are mapping the response curves,
however they obtain them, to minimum phase IIR's.

Do you think they use IIRs to keep latency down? My understanding of such
things is a little vague.



I can't say I've ever seen a graph that came with a mic that contains any
detail whatsoever. They tend to have about a +-10db range and a thick
black line.

And you'll see pretty much the same curve if you extract one
of the Antares impulse responses and transform it. Same
smoothness, same shape as published but no thick black line.
:-)

Having a look at a .amm file in a hex editor, the part that
deals with one pattern of a mic, seems to be about 200-350 samples long
(If they are IRs with 16bit samples, they don't look like it, but the
organisation of the file is quite clear). Is that long enough for an mic
impulse response?

I wonder if the IRs are rather generated 'on the fly' from a splined set
of co-ordinates in that file... That would explain the smoothness.




Based on actual measurement I can make my Sony ECM MS-907 mic sound so
much like a pair of Sound Room MC012 cards in XY that I'd almost be
willing to put them up in ABX. Not quite that confident, though, and
it is well nigh impossible to set up an ABX for microphones. That's
part of what makes it so easy for some to say just about anything they
like about mics and classify their quality by how much they cost.


Move off axis, or work in anything but an anechoic chamber, and all
bets are off.... The patterns won't be the same. Also, put them over a
kit as overheads and I bet they sound totally different, if only
because they will crap out in different ways. (The pads on the
octavas are not very nice). Then try a pair of KM84s instead of the
octavas...

Since real life recording situation variations are at least as great and
usually far greater than the differences in patterns between the
microphones, what I am saying is that what I get out of a transformed
ECM-MS907 is a _very_ plausible MC012 XY.

If you want to loan me a pair of KM84's and a Sony ECM-MS907 for about a
week, I'll send them back to you with IR's that will transform MS907
recordings to a plausible facsimile of KM84 recordings that you can
judge for yourself.

If you decide to take me up on that, I will need a bit of setup time
first to get my new hemi-anechoic measurement rig rung out. Pics he

http://www.arcanemethods.com/Hemi_Anechoic/

The thing will be set up out of doors far from reflections.

Cool. I'd guess that knowing the response of an exact set of
source and modelled mics would work rather better than generic
examples.

Any particular reason for using a ECM-MS907?
Also, what's the driver at the bottom of that strange device?
And... What convolver(s) do you use afterwards? I have Acoustic Mirror and
SIR.

Sadly, I can't loan you a pair of KM84s, as I don't own any.
I tend to work on a beg and borrow basis to supplement my meagre
collection of mics. They do sound nicer than my octavas when I've
got them.

It's an interesting offer though. While I 'know' it should not be possible
to turn an ECM-MS907 into a passable pair of KM84s, I also 'knew' that no
softsynth could ever sound fat like a moog, and that all digital eqs sound
nasty and harsh.


Bob

philicorda wrote:


Do you think they use IIRs to keep latency down? My understanding of such
things is a little vague.

After what you said further down I'm not so sure now that
they are encoded as IIR's but they most definitely are
minimum phase. But latency is the reason for minimum phase.
Of all the ways a filter can match a given magnitude curve,
the minimum phase filter gives the smallest latency.

Having a look at a .amm file in a hex editor, the part that
deals with one pattern of a mic, seems to be about 200-350 samples long
(If they are IRs with 16bit samples, they don't look like it, but the
organisation of the file is quite clear). Is that long enough for an mic
impulse response?

That's a _whole_ lot of coeficients for an IIR. I don't
know of any IIR design algorithms that could calculate that
many, but the IR's that come out of it are certainly longer
than 200-350 samples. When I run a single sample floating
point impulse through it, the resulting IR has much greater
than 16 bit resolution.

I wonder if the IRs are rather generated 'on the fly' from a splined set
of co-ordinates in that file... That would explain the smoothness.

Not sure. They may be encoded some way other than IIRs.
The size you see does sorta indicate that possibility.

If you decide to take me up on that, I will need a bit of setup time
first to get my new hemi-anechoic measurement rig rung out. Pics he

http://www.arcanemethods.com/Hemi_Anechoic/

The thing will be set up out of doors far from reflections.


Cool. I'd guess that knowing the response of an exact set of
source and modelled mics would work rather better than generic
examples.

Always, but I still don't think the ones in MicModeler were
done from actual measurement despite the claims and I think
it can be done generically from measurement with some
averaging over instnces to get much better results. Of
course that presents some very serious aquisition issues
(another reason I don't think all those MicModeler mics were
measured.)


Any particular reason for using a ECM-MS907?

I like it. It's small, self powered, quieter than you would
expect, mid/side, and very inexpensive. It's frequency
response (mag/phase) is not at all good at either end so it
presents an ideal demonstration source mic. It gives a
superb stero image transformed or not. It's all I use now
for live recording because I have the SW I mention below and
have measured a few pretty good mics that I can transform
the recordings to. I can flatten them to the response of
the Earthworks TC30K I use as a "flat" reference mic for my
measurements and sorta like it the best. You can't doubt
the honesty of that transform if you were there when the
recording was made.

I was moving toward a PC SW product and wanted to show what
could be done with the methods to a consumer grade mic when
approached correctly but issues with the programmer I hooked
up with terminated that collaboration. I have a (mostly
working) engineering prototype for the PC (without source
code) but the implementation doesn't really belong to me.

There is a very nicely polished Mac OS X version (screen
shots at my site) done with another, much more civilized and
cooperative, collaborator but sundry individual impediments
arose along the way and momentum was lost. The problem in
the next paragraph is what has me stalled right now. I had
considered just using MicModeler's IR's. I extracted them
and built a working library for each of the two prototypes
but have decided both that the ethics of that are highly
questionable and that they aren't of sufficient quality.

I was also down to the point where building a virtual mic
cabinet for it presented problems I couldn't overcome.
There's no way I can afford to buy an instance of all the
mics that I'd need to have an attractive product and I'm not
well enough known or connected to go borrowing them for a
measurement (and turned around in a day.)

Also, what's the driver at the bottom of that strange
device?

Not sure what you mean.

And... What convolver(s) do you use afterwards? I
have Acoustic Mirror and SIR.

It could be done with either and an appropriate editor. A
more convenient combination than that would be AudioMulch
with SIR. In the product prototypes I have, it is done with
internally implemented overlap-add fast convolution
employing a library FFT.

It's an interesting offer though. While I 'know' it should not be possible
to turn an ECM-MS907 into a passable pair of KM84s, I also 'knew' that no
softsynth could ever sound fat like a moog, and that all digital eqs sound
nasty and harsh.

Precisely. :-)


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

philicorda wrote:


Do you think they use IIRs to keep latency down? My understanding of such
things is a little vague.

After what you said further down I'm not so sure now that
they are encoded as IIR's but they most definitely are
minimum phase. But latency is the reason for minimum phase.
Of all the ways a filter can match a given magnitude curve,
the minimum phase filter gives the smallest latency.

Having a look at a .amm file in a hex editor, the part that
deals with one pattern of a mic, seems to be about 200-350 samples long
(If they are IRs with 16bit samples, they don't look like it, but the
organisation of the file is quite clear). Is that long enough for an mic
impulse response?

That's a _whole_ lot of coeficients for an IIR. I don't
know of any IIR design algorithms that could calculate that
many, but the IR's that come out of it are certainly longer
than 200-350 samples. When I run a single sample floating
point impulse through it, the resulting IR has much greater
than 16 bit resolution.

I wonder if the IRs are rather generated 'on the fly' from a splined set
of co-ordinates in that file... That would explain the smoothness.

Not sure. They may be encoded some way other than IIRs.
The size you see does sorta indicate that possibility.

If you decide to take me up on that, I will need a bit of setup time
first to get my new hemi-anechoic measurement rig rung out. Pics he

http://www.arcanemethods.com/Hemi_Anechoic/

The thing will be set up out of doors far from reflections.


Cool. I'd guess that knowing the response of an exact set of
source and modelled mics would work rather better than generic
examples.

Always, but I still don't think the ones in MicModeler were
done from actual measurement despite the claims and I think
it can be done generically from measurement with some
averaging over instnces to get much better results. Of
course that presents some very serious aquisition issues
(another reason I don't think all those MicModeler mics were
measured.)


Any particular reason for using a ECM-MS907?

I like it. It's small, self powered, quieter than you would
expect, mid/side, and very inexpensive. It's frequency
response (mag/phase) is not at all good at either end so it
presents an ideal demonstration source mic. It gives a
superb stero image transformed or not. It's all I use now
for live recording because I have the SW I mention below and
have measured a few pretty good mics that I can transform
the recordings to. I can flatten them to the response of
the Earthworks TC30K I use as a "flat" reference mic for my
measurements and sorta like it the best. You can't doubt
the honesty of that transform if you were there when the
recording was made.

I was moving toward a PC SW product and wanted to show what
could be done with the methods to a consumer grade mic when
approached correctly but issues with the programmer I hooked
up with terminated that collaboration. I have a (mostly
working) engineering prototype for the PC (without source
code) but the implementation doesn't really belong to me.

There is a very nicely polished Mac OS X version (screen
shots at my site) done with another, much more civilized and
cooperative, collaborator but sundry individual impediments
arose along the way and momentum was lost. The problem in
the next paragraph is what has me stalled right now. I had
considered just using MicModeler's IR's. I extracted them
and built a working library for each of the two prototypes
but have decided both that the ethics of that are highly
questionable and that they aren't of sufficient quality.

I was also down to the point where building a virtual mic
cabinet for it presented problems I couldn't overcome.
There's no way I can afford to buy an instance of all the
mics that I'd need to have an attractive product and I'm not
well enough known or connected to go borrowing them for a
measurement (and turned around in a day.)

Also, what's the driver at the bottom of that strange
device?

Not sure what you mean.

And... What convolver(s) do you use afterwards? I
have Acoustic Mirror and SIR.

It could be done with either and an appropriate editor. A
more convenient combination than that would be AudioMulch
with SIR. In the product prototypes I have, it is done with
internally implemented overlap-add fast convolution
employing a library FFT.

It's an interesting offer though. While I 'know' it should not be possible
to turn an ECM-MS907 into a passable pair of KM84s, I also 'knew' that no
softsynth could ever sound fat like a moog, and that all digital eqs sound
nasty and harsh.

Precisely. :-)


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

In article , says...




philicorda wrote:


Do you think they use IIRs to keep latency down? My understanding of such
things is a little vague.

After what you said further down I'm not so sure now that
they are encoded as IIR's but they most definitely are
minimum phase. But latency is the reason for minimum phase.
Of all the ways a filter can match a given magnitude curve,
the minimum phase filter gives the smallest latency.

Having a look at a .amm file in a hex editor, the part that
deals with one pattern of a mic, seems to be about 200-350 samples long
(If they are IRs with 16bit samples, they don't look like it, but the
organisation of the file is quite clear). Is that long enough for an mic
impulse response?

That's a _whole_ lot of coeficients for an IIR. I don't
know of any IIR design algorithms that could calculate that
many, but the IR's that come out of it are certainly longer
than 200-350 samples. When I run a single sample floating
point impulse through it, the resulting IR has much greater
than 16 bit resolution.

It's really not meaningful to talk about the "length" of the impulse response
of an Infinite Impulse Response filter. Eventually, the impulse response
falls into the noise floor if everything has been properly dithered.
Otherwise, it might suddenly truncate. Or, if the designer hasn't been
careful, it will end up in a limit cycle and continue indefinitely.

You are right, however, that it's virtually unthinkable to have a 200th order
IIR filter unless you are cascading a bunch of allpasses together in an
attempt to closely approximate a given group delay specification (less, of
course, an arbitrary constant delay). And even then, 200 coefficients is a
real stretch.

In article , says...




philicorda wrote:


Do you think they use IIRs to keep latency down? My understanding of such
things is a little vague.

After what you said further down I'm not so sure now that
they are encoded as IIR's but they most definitely are
minimum phase. But latency is the reason for minimum phase.
Of all the ways a filter can match a given magnitude curve,
the minimum phase filter gives the smallest latency.

Having a look at a .amm file in a hex editor, the part that
deals with one pattern of a mic, seems to be about 200-350 samples long
(If they are IRs with 16bit samples, they don't look like it, but the
organisation of the file is quite clear). Is that long enough for an mic
impulse response?

That's a _whole_ lot of coeficients for an IIR. I don't
know of any IIR design algorithms that could calculate that
many, but the IR's that come out of it are certainly longer
than 200-350 samples. When I run a single sample floating
point impulse through it, the resulting IR has much greater
than 16 bit resolution.

It's really not meaningful to talk about the "length" of the impulse response
of an Infinite Impulse Response filter. Eventually, the impulse response
falls into the noise floor if everything has been properly dithered.
Otherwise, it might suddenly truncate. Or, if the designer hasn't been
careful, it will end up in a limit cycle and continue indefinitely.

You are right, however, that it's virtually unthinkable to have a 200th order
IIR filter unless you are cascading a bunch of allpasses together in an
attempt to closely approximate a given group delay specification (less, of
course, an arbitrary constant delay). And even then, 200 coefficients is a
real stretch.

Bob Cain wrote:
philicorda wrote:

Do you think they use IIRs to keep latency down? My understanding of such
things is a little vague.

After what you said further down I'm not so sure now that
they are encoded as IIR's but they most definitely are
minimum phase. But latency is the reason for minimum phase.
Of all the ways a filter can match a given magnitude curve,
the minimum phase filter gives the smallest latency.

If they are minimum-phase, they are not FIR filters.

I have used some digital systems with FIR filters, and they are very strange
to work with. They don't feel like "normal" EQ at all, and using them to
try and undo minimum-phase problems doesn't work very well. But they are
great for notching and slicing, for example.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."

Bob Cain wrote:
philicorda wrote:

Do you think they use IIRs to keep latency down? My understanding of such
things is a little vague.

After what you said further down I'm not so sure now that
they are encoded as IIR's but they most definitely are
minimum phase. But latency is the reason for minimum phase.
Of all the ways a filter can match a given magnitude curve,
the minimum phase filter gives the smallest latency.

If they are minimum-phase, they are not FIR filters.

I have used some digital systems with FIR filters, and they are very strange
to work with. They don't feel like "normal" EQ at all, and using them to
try and undo minimum-phase problems doesn't work very well. But they are
great for notching and slicing, for example.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."

(Mike Rivers) wrote

We have a global economy now. For some people, it may not be a
hardship to pay for shipping from Africa for the the right piece of
gear. eBay is your friend if you're a seller, and occasionally if
you're a buyer.

Alas, I suffer from severe eBay envy. I suppose you're right in
certain cases, but the price of shipping from here to the US is
extortionate -- it's almost as if FedEx doesn't really want the
business. A guitar, for instance, costs hundreds of dollars; I forget
the exact amount, but it is prohibitive.

And there's another wrinkle on this question, too. Because we pay high
import duties on luxury goods (into which category most electronic
stuff falls), driving their prices up, the local second-hand price --
if you can find a buyer -- is considerably higher than what an
American would pay.

Raglan

(Mike Rivers) wrote

We have a global economy now. For some people, it may not be a
hardship to pay for shipping from Africa for the the right piece of
gear. eBay is your friend if you're a seller, and occasionally if
you're a buyer.

Alas, I suffer from severe eBay envy. I suppose you're right in
certain cases, but the price of shipping from here to the US is
extortionate -- it's almost as if FedEx doesn't really want the
business. A guitar, for instance, costs hundreds of dollars; I forget
the exact amount, but it is prohibitive.

And there's another wrinkle on this question, too. Because we pay high
import duties on luxury goods (into which category most electronic
stuff falls), driving their prices up, the local second-hand price --
if you can find a buyer -- is considerably higher than what an
American would pay.

Raglan

Scott Dorsey wrote:

Bob Cain wrote:

philicorda wrote:


Do you think they use IIRs to keep latency down? My understanding of such
things is a little vague.

After what you said further down I'm not so sure now that
they are encoded as IIR's but they most definitely are
minimum phase. But latency is the reason for minimum phase.
Of all the ways a filter can match a given magnitude curve,
the minimum phase filter gives the smallest latency.


If they are minimum-phase, they are not FIR filters.

Any FIR (all zero filter) can be made minimum phase by
reflecting all its z-domain zeros that are outside the unit
circle to the inside. The magnitude response will be
identical and the phase shift will be as low as it can be at
every frequency (for a filter of that length.) If you mean
that there will always be a longer filters that can give the
same magnitude response with less phase shift and that the
minimum minimum phase filter for a given magnitude response
must be infinite then I think you could well be right. I've
never seen that particular discussion come up but it
certainly has a ring of truth to it.

In a strict sense, no IIR can be implemented by an FIR,
obviously, but real world IIRs nearly always have a tail
that is below signifigance within a length easily handled by
an FIR. If you just truncate and window, or often just
truncate, the difference will usually be negligable with
just some fractional DB and group delay differences at the
lowest frequencies.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

Scott Dorsey wrote:

Bob Cain wrote:

philicorda wrote:


Do you think they use IIRs to keep latency down? My understanding of such
things is a little vague.

After what you said further down I'm not so sure now that
they are encoded as IIR's but they most definitely are
minimum phase. But latency is the reason for minimum phase.
Of all the ways a filter can match a given magnitude curve,
the minimum phase filter gives the smallest latency.


If they are minimum-phase, they are not FIR filters.

Any FIR (all zero filter) can be made minimum phase by
reflecting all its z-domain zeros that are outside the unit
circle to the inside. The magnitude response will be
identical and the phase shift will be as low as it can be at
every frequency (for a filter of that length.) If you mean
that there will always be a longer filters that can give the
same magnitude response with less phase shift and that the
minimum minimum phase filter for a given magnitude response
must be infinite then I think you could well be right. I've
never seen that particular discussion come up but it
certainly has a ring of truth to it.

In a strict sense, no IIR can be implemented by an FIR,
obviously, but real world IIRs nearly always have a tail
that is below signifigance within a length easily handled by
an FIR. If you just truncate and window, or often just
truncate, the difference will usually be negligable with
just some fractional DB and group delay differences at the
lowest frequencies.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

"Arny Krueger" wrote in message ...
"Buster Mudd" wrote in message


Maybe that explains why I hate tracking or overdubbing with one
headphone earpiece on & the other off; I'm much more comfortable with
both earpieces halfway on (raised up 1" or so) if I need to hear
acoustic room sound.

I think that could be true, but it's something different - basically Buster
you've criticized and indicted your monitoring mix. There's something in the
room you need that it does not include - perhaps the sound of your voice as
*processed* by the room and your HRTF.

Actually, when I'm singing I rarely take the cans off; it's when I'm
playing bass and/or (especially) when I'm playing with a drummer that
I want to hear direct sound. Yeah, that's an indictment of headphone
monitoring in general, sure. In a perfect world, headphones would have
no place in the tracking room.

On Sat, 04 Sep 2004 16:51:25 GMT, "Bob Olhsson"
wrote:

... There's just as big of an
ego trip available from using the cheapest, "good enough" gear as copping an
attitude that only expensive gear is "good enough" for you.

Of course, there are people to who your "cheapest" is "the best they
could lay their hands onto", or at least "the most expensive they could
afford".

On Sat, 04 Sep 2004 16:51:25 GMT, "Bob Olhsson"
wrote:

... There's just as big of an
ego trip available from using the cheapest, "good enough" gear as copping an
attitude that only expensive gear is "good enough" for you.

Of course, there are people to who your "cheapest" is "the best they
could lay their hands onto", or at least "the most expensive they could
afford".

On 5 Sep 2004 15:30:38 -0700, (Raglan)
wrote:

Telling people you can shoot a cinematica quality movie with your home video
camera is in essence what this thread is trying to do.

Regards

Leon

Leon,

I see what you're getting at but your analogy is extravagant. More
accurately, you might say that I'm telling people you can shoot
broadcast-quality television on a three-CCD prosumer video camera.
Which is more or less true.

And don't forget, there are Lo-Fi film festivals that won't accept
anything but VHS. They started switching to digital media, though.

On 5 Sep 2004 15:30:38 -0700, (Raglan)
wrote:

Telling people you can shoot a cinematica quality movie with your home video
camera is in essence what this thread is trying to do.

Regards

Leon

Leon,

I see what you're getting at but your analogy is extravagant. More
accurately, you might say that I'm telling people you can shoot
broadcast-quality television on a three-CCD prosumer video camera.
Which is more or less true.

And don't forget, there are Lo-Fi film festivals that won't accept
anything but VHS. They started switching to digital media, though.

We have a global economy now.

Which functiones in a way that quality goods always go one way while
worthless dollars go another.

We have a global economy now.

Which functiones in a way that quality goods always go one way while
worthless dollars go another.

On Wed, 8 Sep 2004 15:08:05 +1000, in rec.audio.pro you wrote:

Adobe Premiere or Final cut pro anyway.

Or Vegas. Or plethora of freeware stuff.

On Wed, 8 Sep 2004 15:08:05 +1000, in rec.audio.pro you wrote:

Adobe Premiere or Final cut pro anyway.

Or Vegas. Or plethora of freeware stuff.

On Mon, 06 Sep 2004 22:25:56 -0400, in rec.audio.pro you wrote:

Mike Rivers wrote:

I have known professionals who have issued recordings made on cassette
with inexpensive mics. They didn't have the low lows or the high
highs, but they had what it takes to make a professional recording.

What, in your mind, determines whether a recording is professional or
not?

And I know professionals who have issued recordings on cassette tape
without cassette. You by tape in small plastic bag. If you want to
listen to it, you have to put the tape into empty cassette shell.

On Mon, 06 Sep 2004 22:25:56 -0400, in rec.audio.pro you wrote:

Mike Rivers wrote:

I have known professionals who have issued recordings made on cassette
with inexpensive mics. They didn't have the low lows or the high
highs, but they had what it takes to make a professional recording.

What, in your mind, determines whether a recording is professional or
not?

And I know professionals who have issued recordings on cassette tape
without cassette. You by tape in small plastic bag. If you want to
listen to it, you have to put the tape into empty cassette shell.